VoIP Dial Peers

dial-peer voice 200 voip application session session protocol sipv2 session target sip-server incoming called-number 1222555[4-5].. codecg711ulaw	This peer is currently not being used. Peer 200 will become active once I port my DIDs to the production PoP.
dial-peer voice 201 voip translation-profile outgoing add-ani application session destination-pattern 1[0-9]......... session protocol sipv2 session target sip-server dtmf-relay rtp-nte codec g711ulaw bytes 80 no vad	Peer 201 is used for all domestic calls out from the CME. The "translation-profile outgoing add-ani" is needed to add the additional 7 digits to the front of the callee number in the From line of the SIP messages. For further explanation on why I needed to do this, see telephony-service page. "dtmf-relay rtp-nte" is needed to utilize RFC2833. The "no vad" option is used instead of VAD because enabling VAD can cause a very loud noise on the terminating phone when traveling across third party VoIP networks. The CME side only hears a clicking sound.
dial-peer voice 204 voip application session destination-pattern 011T session protocol sipv2 session target sip-server codec g711ulaw bytes 80 no vad	Peer 204 is for all international dialing. I set up this peer so I can micro manage who needs access to international calls. To allow users to dial international numbers, add "after-hour exempt" command to their ephone config.
dial-peer voice 400 voip description Voice mail dial peer destination-pattern 22[4,9].. session protocol sipv2 session target ipv4:192.168.1.2 dtmf-relay rtp-nte codec g711ulaw	Dial peer 400 allows access to the individual voice mail boxes and VoiceMailMain on Asterisk. Since I use 4 digit extensions, I had to add an additional digit in front of the 4 digit extension to prevent the call from looping around back to itself. Same holds true for the access number to VoiceMailMain. For more information see telephone-service and ephone-dn.